I think this should be a quick fix since it's rendering the latest stable
version useless and making the impression that it was released just to break
things and force people onto 1.8x. Just a thought...no blame game. But
really something like this should be tackled quickly. No point to break
thing
I've used the manager interface to make calls successfully, now I'd like
a look at some of he other ways it can be used.
I've seen references to its use to perform call cut off and rate CDRs.
Is anyone aware of a reference or tutorial I could look at?
Bruce Ferrell
--
_
On 07/29/2011 06:20 PM, Paul Belanger wrote:
On 11-07-29 06:12 PM, Bruce B wrote:
Hi everyone,
Asterisk 1.6.2.19 has a bug per:
https://issues.asterisk.org/jira/browse/ASTERISK-18103
What is the general time to fix this? I think a similar thing is also
noted
in 1.8x install. Is it not going to
On 11-07-29 06:12 PM, Bruce B wrote:
Hi everyone,
Asterisk 1.6.2.19 has a bug per:
https://issues.asterisk.org/jira/browse/ASTERISK-18103
What is the general time to fix this? I think a similar thing is also noted
in 1.8x install. Is it not going to be taken care of because it's 1.6x ?
1.6.2.1
Hi everyone,
Asterisk 1.6.2.19 has a bug per:
https://issues.asterisk.org/jira/browse/ASTERISK-18103
What is the general time to fix this? I think a similar thing is also noted
in 1.8x install. Is it not going to be taken care of because it's 1.6x ?
Thanks
--
> We enable pri intense debug with the standard asterisk PRI dialplan,
> collected the logs and you can find the logs attached to the mail.
>
> After the call was made, the called party cut the call, and asterisk
> doesn't seem to recognise the event.
>
> I can't make much sense of the logs given
Hello,
We enable pri intense debug with the standard asterisk PRI dialplan,
collected the logs and you can find the logs attached to the mail.
After the call was made, the called party cut the call, and asterisk doesn't
seem to recognise the event.
I can't make much sense of the logs given my no
HI Eric,
is a mobile number in India, and the call id rejected by ending the
call from the mobile.
BTW, why is the mail going to asterisk-users-bounces?
--
Thanks,
Ishwar.
On Fri, Jul 29, 2011 at 9:34 PM, Eric Wieling wrote:
>
>
> > -Original Message-
> > From: asterisk-users-boun
Yep. Look the dtails of option of Dial command and features.conf.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinod
Dharashive
Sent: Friday, July 29, 2011 8:51 PM
To: asterisk-users@lists.digium.com
Subject:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ishwar Sridharan
> Sent: Friday, July 29, 2011 9:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Captur
ok I'll do it Monday, and how you handle it with the version 1.10?
Thanks
--
From: "Danny Nicholas"
Sent: Friday, July 29, 2011 5:05 PM
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Subject: Re: [asterisk-users] call forwardin
Hi team,
Is it possible to capture dtmf input once call is patched between a-party and
b-party? Also on dtmf input issue hangup request to b-party with out
disconnecting A-party.
How is this scenario implemented in dialplan?
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
--
___
Thank you Dave.
--
Thanks, Phil
- Original Message -
> On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote:
> > Hi,
> >
> > compiling up a new installation of Asterisk 1.8.5 on CentOS 6
> > X86_64 and
> > am seeing the following when running the make:
> >
> > /usr/bin/ld: skipping incompatible /u
Upgrade to 1.8/10.0
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 10:04 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outs
So I can't do anything?
--
From: "Kevin P. Fleming"
Sent: Friday, July 29, 2011 4:48 PM
To:
Subject: Re: [asterisk-users] call forwarding number from outside.
On 07/29/2011 10:41 AM, Danny Nicholas wrote:
Couple of questions -
This "magic tr
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Friday, July 29, 2011 9:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.
On 07/2
On 07/29/2011 10:41 AM, Danny Nicholas wrote:
Couple of questions -
This "magic trick" is contained in app_dial?
Functionality is inherent to 1.8/10.X structure so we can't re-invent this
in our old 1.4/1.6 installs?
No, it's core functionality, implemented in the channel drivers and
using
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Friday, July 29, 2011 8:49 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.
On 07/2
On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote:
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and
am seeing the following when running the make:
/usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for
-lpam
/usr/bin/ld: skipping incompatible /usr/lib/lib
HI Eric, Nikhil,
Thanks a lot for the responses. Bear with me a little as I'm very new to
asterisk.
I reproduced the problem using standard dialplan. The following are the
configuration files:
*chan_dahdi.conf*
*[trunkgroups]
[channels]
language=en
nationalprefix=+91
pridialplan=national ; or nat
On 07/29/2011 09:12 AM, Eric Wieling wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, July 29, 2011 9:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Danny Nicholas
> Sent: Friday, July 29, 2011 9:06 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Cc: jim.smith...@debsinc.com
> Subjec
The issue with assisted transfer is that the "assisting transferer" is a
second call
Outside -> A
A answers
A calls B to tell them they have a call (call #2 with ID of A
A transfers Outside but the ID stays A
Blind Transfer
Outside -> A
A answers
A blind transfers to B (1 call - keeps ID
- Original Message -
> What you are wanting to do is, in essence, like teaching a gerbil to
> bark.
> It's an extraordinary effort to go to when you can get puppies
> anywhere; and
> at the end of the day, it's still not, and never will be, a proper
> dog.
>
+1 I could not have said it be
On Friday 29 Jul 2011, virendra bhati wrote:
> Hi List,
>
> I want to use these features but nothing was found after googling . please
> give me some examples
>
> Asterisk CLI prompt
> Changing the CLI Prompt
>
> The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable
> that
> you
Use This Information.
You can customize the prompt a bit, if the default prompt is too dull for
you. First add these lines to */etc/asterisk/extensions.conf* in the
[globals] section:
${ENV(UNIX)}
${ENV(ASTERISK_PROMPT)}
Then in */etc/profile* on the Asterisk server, set the ASTERISK_PROMPT
valu
Hi List,
I want to use these features but nothing was found after googling . please
give me some examples
Asterisk CLI prompt
Changing the CLI Prompt
The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable
that
you set from the Unix shell before starting the Asterisk CLI (not th
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am
seeing the following when running the make:
/usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam
/usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl
/usr/bin/l
Voip.ms actually offers more features. Depends on your needs. I use
both as long distance carriers. My DID's are from Voip.ms.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A.H. Jos
Sent: Friday, July 29, 2011 4:53 AM
To: Asterisk
Hello,
Asterisk seems to try to authenticate incoming INVITE based on the [section]
in sip.conf and not the username specified.
I just removed the "insecure" option from my sip.conf requesting every
connection to be authenticated. I added the match_auth_username=yes in the
[general] section for ex
Thank you Terry, CallWithUs is what I am looking for, "the most feature
rich" VoIP service!!!
I hope it will not be difficult for me to have it working with Asterisk and
OpenBTS (It's worth to see what OpenBTS is)
On Thu, Jul 28, 2011 at 12:48 PM, Terry Brummell wrote:
> Yes, they used to allow
On Thursday 28 Jul 2011, Gilles wrote:
> On Thu, 28 Jul 2011 12:04:38 +0500, "Faisal Hanif"
> wrote:
> >I have tried asterisk on windows XP using Cygwin and it worked fine.
> Would you mind explaining how to do this?
I hate to sound patronising but, if you need to ask how to install Cygwin on
Wi
Hi,
One more thing previously there was a project named as AstWin which was
maintaining asterisk's port to windows and providing an installable package
of Asterisk for windows. I am not aware about current state of project but,
I have installation package of Asterisk for windows version 1.2.
I
Did you tried to execute Set(CALLERID(num)=you-required-callerid)?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Friday, July 29, 2011 1:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [a
Hi,
I haven't write any How to on it but below are some step by step
instructions to run Asterisk on windows,
1-Install Cygwin.
2-Install build essentials in Cygwin.
3-Download Asterisk source (I used 1.4.x) and unzip it using tar (You may
need to install tar manually as it is missing in some Cyg
Thanks for the reply!
I've tried and works, but isn't possible with the transfer assisted?
thanks
From: Mike
Sent: Friday, July 29, 2011 8:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] call forwarding number from outside.
That`s the normal
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